[OSL | CCIE_Voice] MRG AND MRGL
Hi to All I am wondering if one site needs both xcoder and MTP then what is the best practice with MRGL Its best to create one MRG and add there both xcoder and MTP or its better to create one MRG for xcoder , one MRG for MTP and then assign both MRG into MRGL What is the choice Regards c c ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] HQ and Branch1 phones cannot call CUE
Hi Williams, Thanks for your email. Earlier message on the calling hq-phones was not enough bandwidth when i tried to place the call to 4220, after reload whole lab now its Ring out and fast busy. When i dial 4 digit extension of BR2 phone from HQ or BR1 phone call goes smooth no prob upto 4 calls after that , not enough bandwidth, Rerouting on display of phone and call goes via PSTN network I think this is expected ok This happens vice versa, sc to hq. I configured sdspfarm transcode session 4 under telephony-service. my configuration is CME-SRST My problem is HQ and BR1 Phones can't call CUE VM Pilot . as mentioned above I am using RSVP, its configures exactly how you have explained working ok. Automated Alternate Routing in Service Parameters = TRUE. HQ and BR2 you need to configure MTP resources = Done configure Location RSVP setting to mandatory between HQ --BR2 = Done I have not configured ip precedence under sccp, does that matters, I have added now but situation remain same. i dont have serial uplink its ethernet only, but works as per my testing voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol cisco ! voice-card 0 dspfarm dsp services dspfarm ! interface Loopback0 ip address 172.16.30.254 255.255.255.0 ip pim dense-mode ip ospf network point-to-point ! interface FastEthernet0/0 ip address 192.168.1.4 255.255.255.0 duplex auto speed auto ip rsvp bandwidth 112 ! interface Service-Engine0/0 ip unnumbered Loopback0 service-module ip address 172.16.30.253 255.255.255.0 service-module ip default-gateway 172.16.30.254 ! interface Vlan31 ip address 172.22.30.254 255.255.255.0 ip pim dense-mode ! interface Vlan32 ip address 172.32.30.254 255.255.255.0 ! ! sccp local Loopback0 sccp ccm 172.20.10.12 identifier 1 priority 1 version 7.0 sccp ccm 172.20.10.11 identifier 2 priority 2 version 7.0 sccp ccm 172.22.30.254 identifier 3 priority 3 version 7.0 sccp ! sccp ccm group 1 bind interface Loopback0 associate ccm 1 priority 1 associate ccm 2 priority 2 associate ccm 3 priority 3 associate profile 3 register sc-conf associate profile 1 register sc-mtp associate profile 2 register sc-xcode ! dspfarm profile 2 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 4 associate application SCCP ! dspfarm profile 1 mtp codec g729r8 codec pass-through rsvp maximum sessions software 8 associate application SCCP ! Regards, Vikky On Thu, Mar 21, 2013 at 7:47 AM, Sears, Michael (msears) michael.se...@compucom.com wrote: Greetings Vikas, First, do you get any message on the calling phone, like not enough bandwidth when you try and place the call. What happens when you try and dial 4 digit extension of BR2 phone from HQ or BR1 phone, Rapid busy, dead air, not enough bandwidth on display of phone, Rerouting? First this could just be a simple case of not configuring sdspfarm transcode session 10 under telephony-service. It all depends on your configuration and if SRST is involved. I just read in your header HQ and BR1 Phones can't call CUE VM Pilot so the following may not help and your problem is local on BR2 router. How are you trying to trigger CAC? Are you in fact trying to configure RSVP based CAC or plain simple locations based CAC? It is unclear what it is your trying to accomplish. If your trying to perform RSVP Based CAC how many calls do you want to permit. Let's say for example you want to permit 4 calls then reroute across the PSTN using AAR. In this case you would need to turn on Automated Alternate Routing in Service Parameters. Then on HQ and BR2 you need to configure MTP resources. In addition you need to configure Location RSVP setting to mandatory between HQ --BR2. You also need to configure on HQ and BR2: interface Serial0/1/0:0.102 point-to-point ip rsvp bandwidth 112 (to allow 4 calls) ! sccp local Loopback0 sccp ccm [ip address] identifier 1 priority 1 version 6.0 sccp ccm [ip address] identifier 2 priority 2 version 6.0 sccp ccm [ip address] identifier 3 priority 3 version 6.0 sccp ip precedence 3 sccp ! sccp ccm group 1 description sccp ccm group 1 bind interface Loopback0 associate ccm 1 priority 1 associate ccm 2 priority 2 associate ccm 3 priority 3 associate profile 1 register sc-mtp associate profile 2 register sc-xcode associate profile 3 register sc-conf registration timeout 3 registration retri 3 keepalive timeout 3 keepalive retri 3 switchback met imm switchback interval 15 switchover met imm ! dspfarm profile 1 mtp description dspfarm profile 1 mtp codec pass-through (I always use codec pass-through some do not it seem to work both ways) codec g729r8 rsvp maximum sessions software 10 associate application SCCP ! dspfarm profile 2 transcode description dspfarm profile 2 transcode !
Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension
Easy way of doing this is to copy the hunt pilot and give it another number.. set user caller ID and mask it to Then in the call-manager-fallback change the voicemail to the new hunt pilot and your done Leslie Meade .. Mobile:778.228.4339 | Main: 604.676.5239 Email: leslie.me...@lvs1.commailto:leslie.me...@lvs1.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Chadi H Hassoune (chassoun) Sent: Thursday, March 21, 2013 7:10 PM To: Pixar Perfect; Mark Thrash (marthras); Steve Keller Cc: CCIE Voice OSL Subject: Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension Calling Party Xform and assign it to the CUC Device Pool works fine for me. HTH From: Pixar Perfect pixarperf...@live.commailto:pixarperf...@live.com Date: Wednesday, March 20, 2013 11:43 PM To: Mark Thrash (marthras) marth...@cisco.commailto:marth...@cisco.com, Steve Keller skeller...@gmail.commailto:skeller...@gmail.com Cc: CCIE Voice OSL ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension the requirement is always for SiteB calling into SiteA voicemail by hitting Messages button. SiteA is always MGCP gateway. Calling Party Xform isnt any use on MGCP gateway for incoming calls. here is another way of doing it ... Voicemail Pilot for CUC is 2200 call-manager-fallback voicemail 2777 --- siteB specific translation-pattern on CUCM to convert 2777 into 2200 and mask calling number . The CSS of the translation pattern should have access to 2200. there is no definitive answer as to which solution is graded positively. there is a reason why many leading CCIE instructors say this is not a test of best practices but a test of how like able is your solution to the script. .. :) From: marth...@cisco.commailto:marth...@cisco.com To: skeller...@gmail.commailto:skeller...@gmail.com Date: Thu, 21 Mar 2013 03:59:48 + CC: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension What about a calling party transform mask on the incoming gateway? Sent from my iPhone On Mar 20, 2013, at 10:43 PM, Steve Keller skeller...@gmail.commailto:skeller...@gmail.com wrote: Thanks Bill, I like this option pretty well as it seems to limit treatment of calls this way to CUC when site B is in SRST mode only. I will try to lab this up tomorrow morning. Question for you, will this only solve my issue of pressing the VM button to access my mailbox to retrieve a message. Meaning when PSTN calls in to site B phone and then gets forward(redirected) to voicemail, I use a dial-peer that provides RDNIS capabilites to route the caller to the correct mailbox and not the opening greeting. So with this would i still want to use the following to get the caller into my mailbox? dial-peer voice 2600 pots destination-pattern 2600 port 0/0/0:23 no digit-strip prefix 202555 ( assuming no LD code at this site ) this is the way i get callers into my mailbox - using RDNIS. On Wed, Mar 20, 2013 at 10:49 PM, William Bell b...@ucguerrilla.commailto:b...@ucguerrilla.com wrote: If you are told National calls must present a 10D ANI AND you are restricted from using an alternate extension in CUC then I do the following. I am not sure whether this would be graded right or wrong On the SRST device (assume basic SRST) call-manager-fallback max-ephone 10 max-dn 20 oct huntst chan 1 voicemail 912025552699 ! or some unused DID on Site A call-forward noan 912025552600 time 20 !assume VM pilot is 2600 call-forward busy 912025552600 time-z whatever time-f whatever date-f whatever call-forward pattern .T ! do your dial-peer work, as needed On CUCM: Create a PT: hq_gw-in_pt Create a CSS: hq_gw_css Assign CSS to hq gateway Either a.) create a translation in hq_gw-in_pt Pattern: 2699 xform ANI: xform DNIS: 2600! as in, redirect to regular VM pilot CSS: your regular HQ phone CSS will do OR b.) create a new hunt pilot in hq_gw-in_pt Pattern: 2699 HL: your VM HL xform ANI: Why would I go this path? 1. We had a requirement that National calls are presented with a 10D ANI in SRST mode. I assume that you would already have a translation-p that handles this bit 2. We can't modify the CUC subscriber. 3. This method doesn't interfere with RDNIS to VM 4. This method doesn't interfere with direct or redirect calls from HQ or SiteC Anyway, that is my 2 cents. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 20, 2013, at 9:33 PM, Bill wrote: Traditionally you would use the alternate extension or a on the pilot. So if you we're denied the ability to use alternate extension for this task but had to use it for another, say allowing
Re: [OSL | CCIE_Voice] MRG AND MRGL
Hi, What i have seen many doing trans coder and mtp in one mrg and mrgmrgl DP in case you need to share resource then you can create mrg and attach to different-different MRGLs thanks Vikky On Fri, Mar 22, 2013 at 10:58 AM, Chrysostomos Christofi ch.christ...@logicom.net wrote: Hi to All ** ** I am wondering if one site needs both xcoder and MTP then what is the best practice with MRGL Its best to create one MRG and add there both xcoder and MTP or its better to create one MRG for xcoder , one MRG for MTP and then assign both MRG into MRGL ** ** What is the choice ** ** ** ** Regards c c ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] LAB IN 1 Hr
Good luck :) On Fri, Mar 22, 2013 at 5:03 PM, Peter Rody peterrodyc...@gmail.com wrote: Hello friends, Ready for the attempt just starting from hotel now its long practice please pray for my attempt. Everyone comes out say i pass its feel so good but i think i might be first guy need your prayers starting from my hotel now. Not able to sleep whole night properly. Feel like on a top of the world now Thanks zillon to all of you. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] HQ and Branch1 phones cannot call CUE
Hi Experts, I got my problem solved buy reloading the br2-config , publisher and subscriber. can you please advise me how to tackle the requirement of reload in CCIE Lab exam this is very important for me because my Presence also donot work unless i restart my whole lab please advice... Regards, Vikky On Fri, Mar 22, 2013 at 4:17 PM, Vikky Kumar vikkyne...@gmail.com wrote: Hi Williams, Thanks for your email. Earlier message on the calling hq-phones was not enough bandwidth when i tried to place the call to 4220, after reload whole lab now its Ring out and fast busy. When i dial 4 digit extension of BR2 phone from HQ or BR1 phone call goes smooth no prob upto 4 calls after that , not enough bandwidth, Rerouting on display of phone and call goes via PSTN network I think this is expected ok This happens vice versa, sc to hq. I configured sdspfarm transcode session 4 under telephony-service. my configuration is CME-SRST My problem is HQ and BR1 Phones can't call CUE VM Pilot . as mentioned above I am using RSVP, its configures exactly how you have explained working ok. Automated Alternate Routing in Service Parameters = TRUE. HQ and BR2 you need to configure MTP resources = Done configure Location RSVP setting to mandatory between HQ --BR2 = Done I have not configured ip precedence under sccp, does that matters, I have added now but situation remain same. i dont have serial uplink its ethernet only, but works as per my testing voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol cisco ! voice-card 0 dspfarm dsp services dspfarm ! interface Loopback0 ip address 172.16.30.254 255.255.255.0 ip pim dense-mode ip ospf network point-to-point ! interface FastEthernet0/0 ip address 192.168.1.4 255.255.255.0 duplex auto speed auto ip rsvp bandwidth 112 ! interface Service-Engine0/0 ip unnumbered Loopback0 service-module ip address 172.16.30.253 255.255.255.0 service-module ip default-gateway 172.16.30.254 ! interface Vlan31 ip address 172.22.30.254 255.255.255.0 ip pim dense-mode ! interface Vlan32 ip address 172.32.30.254 255.255.255.0 ! ! sccp local Loopback0 sccp ccm 172.20.10.12 identifier 1 priority 1 version 7.0 sccp ccm 172.20.10.11 identifier 2 priority 2 version 7.0 sccp ccm 172.22.30.254 identifier 3 priority 3 version 7.0 sccp ! sccp ccm group 1 bind interface Loopback0 associate ccm 1 priority 1 associate ccm 2 priority 2 associate ccm 3 priority 3 associate profile 3 register sc-conf associate profile 1 register sc-mtp associate profile 2 register sc-xcode ! dspfarm profile 2 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 4 associate application SCCP ! dspfarm profile 1 mtp codec g729r8 codec pass-through rsvp maximum sessions software 8 associate application SCCP ! Regards, Vikky On Thu, Mar 21, 2013 at 7:47 AM, Sears, Michael (msears) michael.se...@compucom.com wrote: Greetings Vikas, First, do you get any message on the calling phone, like not enough bandwidth when you try and place the call. What happens when you try and dial 4 digit extension of BR2 phone from HQ or BR1 phone, Rapid busy, dead air, not enough bandwidth on display of phone, Rerouting? First this could just be a simple case of not configuring sdspfarm transcode session 10 under telephony-service. It all depends on your configuration and if SRST is involved. I just read in your header HQ and BR1 Phones can't call CUE VM Pilot so the following may not help and your problem is local on BR2 router. How are you trying to trigger CAC? Are you in fact trying to configure RSVP based CAC or plain simple locations based CAC? It is unclear what it is your trying to accomplish. If your trying to perform RSVP Based CAC how many calls do you want to permit. Let's say for example you want to permit 4 calls then reroute across the PSTN using AAR. In this case you would need to turn on Automated Alternate Routing in Service Parameters. Then on HQ and BR2 you need to configure MTP resources. In addition you need to configure Location RSVP setting to mandatory between HQ --BR2. You also need to configure on HQ and BR2: interface Serial0/1/0:0.102 point-to-point ip rsvp bandwidth 112 (to allow 4 calls) ! sccp local Loopback0 sccp ccm [ip address] identifier 1 priority 1 version 6.0 sccp ccm [ip address] identifier 2 priority 2 version 6.0 sccp ccm [ip address] identifier 3 priority 3 version 6.0 sccp ip precedence 3 sccp ! sccp ccm group 1 description sccp ccm group 1 bind interface Loopback0 associate ccm 1 priority 1 associate ccm 2 priority 2 associate ccm 3 priority 3 associate profile 1 register sc-mtp associate profile 2 register sc-xcode associate profile 3 register sc-conf
[OSL | CCIE_Voice] ISDN Interface won't to be bind with ccm-manager
Hi experts, I am trying to configure Branch2-R3 as MGCP VG, and I configured an interface E1. I have problem when try to bind-l3 in the serial interface s0/0/0:15 with ccm-manager , the only option appear is q931??? the gateway won't to register with ccm.. any idea ??? Appreciate your help! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] ISDN Interface won't to be bind with ccm-manager
Dears, The problem was with this command : controller e1 0/0/0 pri-group timeslots 1-12 where it should be controller e1 0/0/0 pri-group timeslots 1-12 service mgcp Thanks On Fri, Mar 22, 2013 at 9:57 PM, CCIEing aboaz...@gmail.com wrote: Hi experts, I am trying to configure Branch2-R3 as MGCP VG, and I configured an interface E1. I have problem when try to bind-l3 in the serial interface s0/0/0:15 with ccm-manager , the only option appear is q931??? the gateway won't to register with ccm.. any idea ??? Appreciate your help! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] HQ and Branch1 phones cannot call CUE.
Greetings Vikas, Can you call 4101 from either HQ or SB or do you get a rapid busy or if you call from HQ to SB to 4101 and no other calls up does the call reroute over the PSTN. If you get rapid busy or if your call immediately reroutes over the PSTN that isn't right. You could have a locations issue from what you explain I'm a little confused. Also it appear that you haven't put into place a rsvp bandwidth statement which is required to perform rsvp calls. On HQ and SC need the following statements: interface Serial0/1/0:0.102 point-to-point ip rsvp bandwidth 112 (to allow 4 calls) Michael Sears Compucom Systems Western Region Senior Infrastructure Solution Consulting Office: +1.720.344.6833 Mobile: +1.303.328.5590 Fax: +1.978.863.0740 -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Friday, March 22, 2013 7:34 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 85, Issue 78 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: HQ and Branch1 phones cannot call CUE (Vikky Kumar) 2. Re: SRST to voicemail without Alternate Extension (Leslie Meade) -- Message: 1 Date: Fri, 22 Mar 2013 16:17:09 +0300 From: Vikky Kumar vikkyne...@gmail.com To: Sears, Michael (msears) michael.se...@compucom.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] HQ and Branch1 phones cannot call CUE Message-ID: ca+4dtjfua+8z2fgff9ajxulz5n5zfalxwez9kiyuigyty3g...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi Williams, Thanks for your email. Earlier message on the calling hq-phones was not enough bandwidth when i tried to place the call to 4220, after reload whole lab now its Ring out and fast busy. When i dial 4 digit extension of BR2 phone from HQ or BR1 phone call goes smooth no prob upto 4 calls after that , not enough bandwidth, Rerouting on display of phone and call goes via PSTN network I think this is expected ok This happens vice versa, sc to hq. I configured sdspfarm transcode session 4 under telephony-service. my configuration is CME-SRST My problem is HQ and BR1 Phones can't call CUE VM Pilot . as mentioned above I am using RSVP, its configures exactly how you have explained working ok. Automated Alternate Routing in Service Parameters = TRUE. HQ and BR2 you need to configure MTP resources = Done configure Location RSVP setting to mandatory between HQ --BR2 = Done I have not configured ip precedence under sccp, does that matters, I have added now but situation remain same. i dont have serial uplink its ethernet only, but works as per my testing voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol cisco ! voice-card 0 dspfarm dsp services dspfarm ! interface Loopback0 ip address 172.16.30.254 255.255.255.0 ip pim dense-mode ip ospf network point-to-point ! interface FastEthernet0/0 ip address 192.168.1.4 255.255.255.0 duplex auto speed auto ip rsvp bandwidth 112 ! interface Service-Engine0/0 ip unnumbered Loopback0 service-module ip address 172.16.30.253 255.255.255.0 service-module ip default-gateway 172.16.30.254 ! interface Vlan31 ip address 172.22.30.254 255.255.255.0 ip pim dense-mode ! interface Vlan32 ip address 172.32.30.254 255.255.255.0 ! ! sccp local Loopback0 sccp ccm 172.20.10.12 identifier 1 priority 1 version 7.0 sccp ccm 172.20.10.11 identifier 2 priority 2 version 7.0 sccp ccm 172.22.30.254 identifier 3 priority 3 version 7.0 sccp ! sccp ccm group 1 bind interface Loopback0 associate ccm 1 priority 1 associate ccm 2 priority 2 associate ccm 3 priority 3 associate profile 3 register sc-conf associate profile 1 register sc-mtp associate profile 2 register sc-xcode ! dspfarm profile 2 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 4 associate application SCCP ! dspfarm profile 1 mtp codec g729r8 codec pass-through rsvp maximum sessions software 8 associate application SCCP ! Regards, Vikky On Thu, Mar 21, 2013 at 7:47 AM, Sears, Michael (msears) michael.se...@compucom.com wrote: Greetings Vikas, First, do you get any message on the calling phone, like not enough bandwidth when
[OSL | CCIE_Voice] Outgoing Calls via T1 failed
Hi all, After configuring my HQ GW as MGCP, then configure my T1 to register with cucm , I was govern by the lack of the DSP resources, which force me to define only 16 channel out of the 23 on my pri-group under the T1 controller configuration !! here is the config : controller T1 0/0/0 pri-group timeslots 1-16 service mgcp Then I faced a problem with my outgoing calls , the calls was dropping due to the cause *Requested circuit/channel not available* My Question here, as there is 16 channel in my Pri-group are already configured, why all calls get dropped with cause of non availability of the resources, Why not to use one of the available channels (1-16) Appreciate your help here is below the output of debug q931 for one of my outgoing calls: *SDN Se0/0/0:23 Q931: TX - SETUP pd = 8 callref = 0x000F * *Bearer Capability i = 0x8090A2 * *Standard = CCITT * *Transfer Capability = Speech * *Transfer Mode = Circuit * *Transfer Rate = 64 kbit/s * *Channel ID i = 0xA98390 * *Exclusive, Channel 16 * *Display i = 'HQ PH1' * *Calling Party Number i = 0x2181, '7772022001' * *Plan:ISDN, Type:National * *Called Party Number i = 0x81, '911' * *Plan:ISDN, Type:Unknown* *ISDN Se0/0/0:23 Q931: RX - RELEASE_COMP pd = 8 callref = 0x800F * *Cause i = 0x82AC1810 - Requested circuit/channel not available* ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Outgoing Calls via T1 failed
Is your gateway registered in CUCM? Are you getting the proper output of your show commands? Show isdn status, show ccm Do you have int seri x/x/x Isdn bind-l3 ccm Did you try no MGCP MGCP? Can you post more of your config? Sent from my iPad On Mar 22, 2013, at 4:52 PM, CCIEing aboaz...@gmail.com wrote: Hi all, After configuring my HQ GW as MGCP, then configure my T1 to register with cucm , I was govern by the lack of the DSP resources, which force me to define only 16 channel out of the 23 on my pri-group under the T1 controller configuration !! here is the config : controller T1 0/0/0 pri-group timeslots 1-16 service mgcp Then I faced a problem with my outgoing calls , the calls was dropping due to the cause Requested circuit/channel not available My Question here, as there is 16 channel in my Pri-group are already configured, why all calls get dropped with cause of non availability of the resources, Why not to use one of the available channels (1-16) Appreciate your help here is below the output of debug q931 for one of my outgoing calls: SDN Se0/0/0:23 Q931: TX - SETUP pd = 8 callref = 0x000F Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98390 Exclusive, Channel 16 Display i = 'HQ PH1' Calling Party Number i = 0x2181, '7772022001' Plan:ISDN, Type:National Called Party Number i = 0x81, '911' Plan:ISDN, Type:Unknown ISDN Se0/0/0:23 Q931: RX - RELEASE_COMP pd = 8 callref = 0x800F Cause i = 0x82AC1810 - Requested circuit/channel not available ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Outgoing Calls via T1 failed
Your debug output has a few clues...but I can't recall offhand if channel 16 in that debug starts at 1 (meaning this is the 16th channel) or 0 (meaning this is the 17th channel). Do inbound calls from pstn work? If yes, its more likely the second option. In the first case, it would appear your issue is on the pstn side. Run show ISDN status and layer 2 should show multiple frame established and layer 3 should show ccm-manager (or similar). If however layer 2 shows tei assigned try the following: Mgcp bind media source lo0 Mgcp bind control source lo0 (Paste those commands twice) Int s0/0/0 No ISDN bind-l3 ccm ISDN bind-l3 ccm No mgcp Mgcp Show ISDN status (Ensure you see multi frame established) Also type show ccm and ensure the gw is registered to your cucm. If not, make sure that your hostname on the router matches what you have in cucm. If you have IP domain-name ipexpert.com in your config then you need to use the fqdn in cucm, such as r3.ipexpert.com. however if you don't have a domain name on the router then you should just have the routers hostname w/o domain such as r3. Now, for the other situation if channel 16 in the debug is really channel 17, that could be caused by using the ccm config command. With this, every time in cucm you reset the mgcp gw it will apply a no mgcp then mgcp and download the config from cucm to the router (and configure a FULL PRI). Ccm config command doesn't work with a fractional PRI, but you could use it to download all the commands, then no ccm config and change the controller commands to use timeslots 1-16 rather than 1-24. (Need to shut voice port, shut int s0/0/0 and no ISDN bind-l3 ccm, shut controller and remove timeslots command, then apply commands in reverse order using fractional timeslots). Not a solution, but just for reference, the default channel order for mgcp PRI is bottom up. If your issue is the latter (ccm config downloaded a full PRI config) and you were set to use ascending channels you would not have seen this issue until the 17th call came from pstn...in real lab you would lose points for having a full PRI I stead of fractional, even if calls did work. The point here is make sure you remove ccm config if you have a fractional PRI. Hope this helps... Justin On Mar 22, 2013 7:36 PM, Bill whl...@gmail.com wrote: Is your gateway registered in CUCM? Are you getting the proper output of your show commands? Show isdn status, show ccm Do you have int seri x/x/x Isdn bind-l3 ccm Did you try no MGCP MGCP? Can you post more of your config? Sent from my iPad On Mar 22, 2013, at 4:52 PM, CCIEing aboaz...@gmail.com wrote: Hi all, After configuring my HQ GW as MGCP, then configure my T1 to register with cucm , I was govern by the lack of the DSP resources, which force me to define only 16 channel out of the 23 on my pri-group under the T1 controller configuration !! here is the config : controller T1 0/0/0 pri-group timeslots 1-16 service mgcp Then I faced a problem with my outgoing calls , the calls was dropping due to the cause *Requested circuit/channel not available* My Question here, as there is 16 channel in my Pri-group are already configured, why all calls get dropped with cause of non availability of the resources, Why not to use one of the available channels (1-16) Appreciate your help here is below the output of debug q931 for one of my outgoing calls: *SDN Se0/0/0:23 Q931: TX - SETUP pd = 8 callref = 0x000F * *Bearer Capability i = 0x8090A2 * *Standard = CCITT * *Transfer Capability = Speech * *Transfer Mode = Circuit * *Transfer Rate = 64 kbit/s * *Channel ID i = 0xA98390 * *Exclusive, Channel 16 * *Display i = 'HQ PH1' * *Calling Party Number i = 0x2181, '7772022001' * *Plan:ISDN, Type:National * *Called Party Number i = 0x81, '911' * *Plan:ISDN, Type:Unknown* *ISDN Se0/0/0:23 Q931: RX - RELEASE_COMP pd = 8 callref = 0x800F * *Cause i = 0x82AC1810 - Requested circuit/channel not available* ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
